Pimjai Wesnarat
2006-Apr-11 07:33 UTC
[Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
Hi, I still cant dial out on Zap and I really have no clue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten => 111,n,Dial(Zap/G1/002212601574) exten => 111,n,NoOp(${DIALSTATUS}) exten => 111,n,Busy() exten => 111,n,Hangup() My zapata.conf is like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no echocancel=yes echocancelwhenbridged=yes ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=>1-15,17-31 And I've got this on my CLI: -- Accepting overlap call from '2212601571' to '111' on channel 0/31, span 1 -- Starting simple switch on 'Zap/31-1' -- Executing Answer("Zap/31-1", "") in new stack -- Executing Ringing("Zap/31-1", "") in new stack -- Executing Wait("Zap/31-1", "2") in new stack -- Executing AbsoluteTimeout("Zap/31-1", "30") in new stack -- Set Absolute Timeout to 30 -- Executing Dial("Zap/31-1", "Zap/G1/002212601574") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/002212601574 -- Moving call from channel 31 to channel 30 Apr 11 16:27:06 WARNING[10322]: chan_zap.c:7745 pri_fixup_principle: Can't fix up channel from 31 to 30 because 30 is already in use Apr 11 16:27:06 WARNING[10322]: chan_zap.c:9046 pri_dchannel: Unable to move channel 30! -- Channel 0/30, span 1 got hangup request Apr 11 16:27:06 WARNING[10966]: app_dial.c:706 wait_for_answer: Unable to forward voice -- Hungup 'Zap/30-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp("Zap/31-1", "CHANUNAVAIL") in new stack -- Executing Busy("Zap/31-1", "") in new stack -- Channel 0/31, span 1 got hangup request == Spawn extension (from-pstn, 111, 7) exited non-zero on 'Zap/31-1' -- Executing NoOp("Zap/31-1", "") in new stack -- Executing Goto("Zap/31-1", "999") in new stack -- Goto (from-pstn,h,999) -- Hungup 'Zap/31-1' Could somebody give me a clue? Pim
Lee Archer
2006-Apr-11 07:44 UTC
[Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
When you find out what's causing it can you let me know as I have 1 system that gets this error and the telco tells me everything is fine with their equipment. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Pimjai Wesnarat Sent: 11 April 2006 15:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use?? Hi, I still cant dial out on Zap and I really have no clue why. I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card 4 ports, 31 channels each and able to receive incoming calls and fax perfectly. I've done this in my dial plan. exten => 111,1,Answer() exten => 111,n,Ringing() exten => 111,n,Wait(2) exten => 111,n,AbsoluteTimeout(30) exten => 111,n,Dial(Zap/G1/002212601574) exten => 111,n,NoOp(${DIALSTATUS}) exten => 111,n,Busy() exten => 111,n,Hangup() My zapata.conf is like this [channels] context=from-pstn group=0 switchtype=euroisdn overlapdial=yes faxdetect=no echocancel=yes echocancelwhenbridged=yes ; PRI port 1 (E1) ; context=1 group=1 signalling=pri_cpe channel=>1-15,17-31 And I've got this on my CLI: -- Accepting overlap call from '2212601571' to '111' on channel 0/31, span 1 -- Starting simple switch on 'Zap/31-1' -- Executing Answer("Zap/31-1", "") in new stack -- Executing Ringing("Zap/31-1", "") in new stack -- Executing Wait("Zap/31-1", "2") in new stack -- Executing AbsoluteTimeout("Zap/31-1", "30") in new stack -- Set Absolute Timeout to 30 -- Executing Dial("Zap/31-1", "Zap/G1/002212601574") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/002212601574 -- Moving call from channel 31 to channel 30 Apr 11 16:27:06 WARNING[10322]: chan_zap.c:7745 pri_fixup_principle: Can't fix up channel from 31 to 30 because 30 is already in use Apr 11 16:27:06 WARNING[10322]: chan_zap.c:9046 pri_dchannel: Unable to move channel 30! -- Channel 0/30, span 1 got hangup request Apr 11 16:27:06 WARNING[10966]: app_dial.c:706 wait_for_answer: Unable to forward voice -- Hungup 'Zap/30-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp("Zap/31-1", "CHANUNAVAIL") in new stack -- Executing Busy("Zap/31-1", "") in new stack -- Channel 0/31, span 1 got hangup request == Spawn extension (from-pstn, 111, 7) exited non-zero on 'Zap/31-1' -- Executing NoOp("Zap/31-1", "") in new stack -- Executing Goto("Zap/31-1", "999") in new stack -- Goto (from-pstn,h,999) -- Hungup 'Zap/31-1' Could somebody give me a clue? Pim _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/
Tim Panton
2006-Apr-11 15:03 UTC
[Asterisk-Users] Dial out on Zap: Can't fix up channel from 31 to 30 because 30 is already in use??
On 11 Apr 2006, at 15:33, Pimjai Wesnarat wrote:> Hi, > > I still cant dial out on Zap and I really have no clue why. > I'm using Asterisk Asterisk 1.2.6 and Zaptel 1.2.5 with Digium card > 4 ports, 31 channels each and able to receive incoming calls and > fax perfectly. > > I've done this in my dial plan. > > exten => 111,1,Answer() > exten => 111,n,Ringing() > exten => 111,n,Wait(2) > exten => 111,n,AbsoluteTimeout(30) > exten => 111,n,Dial(Zap/G1/002212601574)You might want to make that exten => 111,n,Dial(Zap/g1/002212601574) The lowercase g makes it start looking for a free channel from the bottom. Generally your PTT will do the opposite when sending a call (i.e. start at the top), but you can ask them to apply either policy. Tim Panton tim@mexuar.com
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